The present invention generally relates to the field of digital speech communications systems. More specifically, the present invention pertains to an improved and efficient method of providing error detection capabilities in a sub-band coder without significantly increasing the required bit rate over the channel.
Recently, there has been a significant interest in the use of digital speech transmission in land-mobile radio channels. The use of digitally-encoded speech provides the advantages of: facilitating voice privacy/security measures such as encryption; maintaining compatibility with data transmission equipment; permitting the use of modern transmission techniques such as optical communications; facilitating channel-reuse techniques for spectral efficiency; and improving speech quality in adverse environments through the use of digital error protection codes.
Those skilled in the RF communications field appreciate the unique problems faced in providing digital communications over an RF channel. Rapid multipath fading, commonly experienced in high frequency radiotelephone communications, causes significant amplitude and phase changes in the digital bit stream such that data carried by the radio channel becomes garbled and missing. Additionally, co-channel interference, ignition noise, and landline telephone network impairments further degrade the signal quality in a land-mobile radio communications system. Moreover, the audio quality of digitally-encoded speech must be maximized while at the same time maintaining reasonable data rates, system complexity, and equipment costs.
Known techniques for maintaining high quality speech though error-prone channels typically involve significant tradeoffs. For example, the simplest way to correct channel errors is to re-send the original signal upon request by the receiver, e.g., as presently done in Automatic Repeat Request (ARQ) protocols of data communications systems. This method, however, precludes implementation in real time because of the time delays necessary to implement the protocol. Alternatively, error detection and correction may be provided for all of the bits in the data stream through the use of redundancy, as is commonly done in cellular signalling. The addition of redundancy necessitates increasing the data rate, by perhaps a factor of two, to accommodate the error protection coding. However, today's channel spacing specifications for mobile radio systems preclude the use of data rates above 16 kilobits-per-second (kbps).
Several speech coding techniques have been suggested for data rates between 9.6 kbps and 16 kbps. These techniques include adaptive sub-band excited transform coding, continuously variable slope delta (CVSD) modulation, adaptive transform coding, multi-pulse linear predictive coding (LPC), vector-excited LPC, and sub-band coding (SBC). For many of these coding schemes, the speech quality at a low data rate may not be acceptable for use with land-mobile radio systems. Other coding schemes reduce the bit rate through the use of vector quantization of the band energies via the use of a codebook. However, the additional computations required to search the codebook introduce additional delay and complexity into the system. Hence, a tradeoff between system complexity, data rate, and speech quality must be made.
A need, therefore, exists to provide a method for maintaining the accuracy of digitally-encoded speech using a minimum amount of added error protection coding so as to avoid any significant increase in data rate.